-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3275/#review11023
-----------------------------------------------------------

Ship it!



/branches/11/res/res_rtp_asterisk.c
<https://reviewboard.asterisk.org/r/3275/#comment20658>

    This will crash and left_candidate will never be non-NULL.


Take out that ao2_ref and this is good to go.

- Joshua Colp


On March 3, 2014, 5:01 p.m., Jonathan Rose wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3275/
> -----------------------------------------------------------
> 
> (Updated March 3, 2014, 5:01 p.m.)
> 
> 
> Review request for Asterisk Developers, Joshua Colp, Kevin Harwell, and Matt 
> Jordan.
> 
> 
> Bugs: ASTERISK-22911
>     https://issues.asterisk.org/jira/browse/ASTERISK-22911
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> This patch provides a fix for the hold problem by doing the following:
> 
> Once an ICE session is marked as started, we start adding any new remote 
> candidates into a separate list until we get another attempt to start the ICE 
> session.
> Once a call to start the ice session is made, instead of immediately quitting 
> if the session is already started, we check for a difference in the two 
> candidates lists.  If the lists are identical, we wipe out the new list and 
> keep the old one and just quit then going on with the current ICE session. If 
> the lists are changed, we toss the old list and adopt the new one and restart 
> the ICE session.
> 
> 
> Diffs
> -----
> 
>   /branches/11/res/res_rtp_asterisk.c 409155 
> 
> Diff: https://reviewboard.asterisk.org/r/3275/diff/
> 
> 
> Testing
> -------
> 
> SIPML client to Asterisk to Desk Phone
> SIPML calls desk phone
> audio test, got two way audio
> SIPML holds call
> SIPML resumes call
> audio test, got two way audio (previously this would cause one way audio from 
> the SIPML client to the desk phone)
> 
> 
> Thanks,
> 
> Jonathan Rose
> 
>

-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Reply via email to